Analysis and Design of Telecommunications
Systems
Marc Van Droogenbroeck
Montefiore Institute, University of Li`ege, Belgium
Academic year: 2020-2021
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Practical information
I Instructor: Marc Van Droogenbroeck
I Assistant: Renaud Vandeghen
I Evaluation
written(!): theory + exercise
I Slides:
http://orbi.uliege.be
https://orbi.uliege.be/handle/2268/170889
I New from September 2019: manual of exercises: http://hdl.handle.net/2268/239453
General overview
In this course:
1 Theory on signals and noise
2 Study of specific aspects of telecommunications systems 3 Towards engineering rules
But why?
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Components of a telecommunication system
Main components:
1 signals
2 electronics (transmitter, receiver, connectors, repeaters, etc). 3 channel: cable (+ adapters), wireless
4 propagation issues: antennas, fading
Main constraints:
1 shared channel bandwidth
2 perturbations due to noise.
Goals of a good design:
increase the Signal to Noise ratio NS to reach the maximal channel capacity (typically for a white Gaussian noise)
enable a communication with many users (seen as interference for the main link)
3 power consumption
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Main concerns related to signals
I Signal source handling (preparation of the signal, at the source, in the transmitter):
filtering (remove what is useless for communications) analog ↔ digital (digitization)
remove the redundancy in the signal: compression
I Signal over the channel:
signal shaping to make it suitable for transmission (coding, modulation, multiplexing, etc)
signal power versus the noise signal (protect the signal against noise effects)
Outline
1 Reminder
2 Representation of bandpass signals
3 Noise in telecommunications systems
4 Digital modulation
5 Spread spectrum
6 Channels for digital communications and intersymbol
interference
7 Navigation systems
8 Multiplexing
9 Telephone traffic engineering
10 Transmission over twisted pairs (fixed telephone network)
11 Radio engineering
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Outline
1 Reminder
2 Representation of bandpass signals
3 Noise in telecommunications systems
4 Digital modulation
5 Spread spectrum
6 Channels for digital communications and intersymbol
interference
7 Navigation systems
8 Multiplexing
9 Telephone traffic engineering
10 Transmission over twisted pairs (fixed telephone network)
Reminder
I Physical layer (not all the transmission protocols)
I Deterministic - stochastic processes
I A tool for characterizing stochastic processes (including noise) and linear systems: the power spectrum or power spectral density
I Properties of the power spectrum
I Gaussian process I Noise I Modulation I Digital communications 9 / 495
Network
⇒ protocols I
Network
⇒ protocols II
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Deterministic vs stochastic tools
transmitter receiver User’s signal deterministic stochastic
Noise and interference stochastic stochastic
deterministic stochastic signal to consider voltage / current power
power analysis instantaneous power Power Spectral Density (PSD) p(t) = |v(t)|R 2 = R |i(t)|2 EX2(t) =R+∞
Deterministic vs stochastic tools
transmitter receiver User’s signal deterministic stochastic
Noise and interference stochastic stochastic
deterministic stochastic
signal to consider voltage / current power
power analysis instantaneous power Power Spectral Density (PSD) p(t) = |v(t)|R 2 = R |i(t)|2 EX2(t) =R−∞+∞γX(f )df
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More about the
P
ower
S
pectral
D
ensity (
PSD
) I
Let X (t) be a (wide-sense) stationary stochastic process (note that we use a “capital” letter X for stochastic processes).
Definition (Auto-correlation function)
ΓXX (τ ) = E {X (t + τ)X (t)} ∀t (1)
Extremely useful because it expresses the average power (when
τ = 0):
ΓXX (τ = 0) = E n
X2(t)o (2)
In practice, we have that the power PX of a stochastic process is
given by: PX = E n X2(t)o (3)
More about the
P
ower
S
pectral
D
ensity (
PSD
) II
Definition (Power spectrum or power spectral density of a stationary process PSD) γX(f ) = Z +∞ −∞ ΓXX (τ ) e −2πjf τdτ (4) Summary: PX = E n X2(t)o (5) = E {X (t+0)X (t)} = Z +∞ −∞ γX(f )e 2πjf0df (6) = Z +∞ −∞ γX(f )df (7)
Therefore, γX(f ) expresses the distribution of power for all the
frequencies.
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Example of a PSD
Let us consider a signal with a random phase θ, uniformly distributed over [−π, +π] (or, likewise, [0, 2π]) X (t) = Ac cos (2πfct +Θ) (8) 1 Mean of X (t)? µX(t) = E {X (t)} = Z +π −π Accos (2πfct +θ) 1 2π dθ = 0 (9) 2 Auto-correlation? ΓXX (t1, t2) = E {X (t1)X (t2)} (10) = Z +π −π Accos (2πfct1 +θ) Accos (2πfct2 +θ) 1 2π dθ = A 2 c 2 cos [2πfc(t2 − t1)] = A2c 2 cos [2πfcτ ] (11) The signal is stationary. So,
I Power spectral density?
γX(f )= A
2 c
Practical link between x(t) and X (t)? I
Let:
I X (t) be the stochastic (unknown) process behind the scenes,
I x(t) be the observation. This is what you measure/observe. You can estimate the PSD of X (t) by taking the Fourier transform of the observation x(t).
Proof.
Assume x(t) is deterministic and has finite energy, that is
Z +∞ −∞
|x(t)|2dt (13)
Let us define:
I a “pseudo” auto-correlation function by Γxx(τ ) =
Z +∞ −∞
x(t)x(t + τ )dt (14)
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Practical link between x(t) and X (t)? II
I and a “pseudo” PSD, which is an estimate of the true PSD, as
γx(f ) =
Z +∞ −∞
Practical link between x(t) and X (t)? III
Some calculations lead to:
γx(f ) = Z +∞ −∞ Γxx (τ )e−2πjf τdτ (16) = Z +∞ −∞ Z +∞ −∞ x(t)x(t + τ )dt e−2πjf τdτ (17) = Z +∞ −∞ x(t) Z +∞ −∞ x(t + τ )e−2πjf τdτ dt (18) = Z +∞ −∞ x(t) X (f )e2πjftdt (19) = X (f ) Z +∞ −∞ x(t)e2πjftdt (20) = X (f )X∗(f ) (21) = kX (f )k2 (22) 19 / 495
Practical link between x(t) and X (t)? IV
In decibels (remember that v ↔ 10 log10(v ) [dB]),
γx(f ) [dB] = 20 log10kX (f )k (23) So, γX(f ) [dB] can be estimated with the help of kX (f )k.
However,
I γX(f ) is not based on a single observation and therefore
γX(f ) is not equal to γx(f ). x [W ] 10 log10(x) [dBW ] 1 [W ] 0 [dBW ] 2 [W ] 3 [dBW ] 0, 5 [W ] −3 [dBW ] 5 [W ] 7 [dBW ] 10n[W ] 10× n [dBW ]
What if a process is not stationary?
Stationary (in the wide sense) means1 the mean of X (t) is time independent:
µX(t) = µX ∀t (24)
2 the autocorrelation only depends on time differences (and not the origin of time):
ΓXX (t1, t2) = ΓXX (t2 − t1) ∀t1, t2 (25)
If a process X (t) is not stationary, then
I you cannot define a PSD for a non-stationary process.
I But, you can sometimes make it stationary. 2 classical ways worth trying:
1 inject a random (independent) phase, suited for modulated signals:
S(t) = X (t) cos(2πfct +Φ) with pdfΦ(φ) = 2π1 for φ∈ [0, 2π].
2 inject a random (independent) time shift (≡jitter), suited for digital
signals: X (t +T0) with pdfT0(t0) = T1 for t0 ∈ [0, T ].
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Power spectral density and linear systems (= filtering)
Consider a stationary process X (t), a linear system whose transfer function is given by H(f ), and Y (t) the output process.
Theorem (Mean of a filtered stochastic process)
Mean of Y (t):
µY = µXH(0) (26)
Theorem (Wiener-Kintchine)
Power spectrum of a filtered stochastic process Y (t):
Useful considerations about the PSDs
I Sum of (stationary) stochastic processes:
Y (t) = K (t) + N(t) (28) If both signals are uncorrelated (which they are if they are independent), then
γYY(f ) = γKK(f ) + γNN(f ) (29)
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Gaussian processes
What does it mean that a process X (t) is “Gaussian”?
I the probability density function (pdf) of its voltage/current is Gaussian distributed: pdfX = fX(x) = 1 σX√2πe −(x−µX) 2 2σ2 X (30)
I the mean and variance of X suffice to characterize it.
I it is a good approximation for the sum of a number of
independent random variables with arbitrary one-dimensional pdfs.
Useful properties:
I If the input of a linear system is a Gaussian stochastic process, then the output is also a Gaussian process.
Noise - white noise I
Definition (White noise)
A white noise is defined as a stochastic process whose power spectral density is constant for each frequency
γN(f ) = N0 2 W Hz (31) In practice, there is no “pure” white noise, but it does not matter as long as the PSD is constant inside the useful bandwidth.
A common signal in telecommunications is a wide-sense stationary zero-mean white Gaussian noise:
I the probability density function of the voltage of the noise is a Gaussian.
I the observed mean voltage has a zero mean.
I its power spectrum is constant for each frequency.
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Noise - white noise II
Fact
Power of a white noise (for a B large bandwidth) PN = N = Z +∞ −∞ γN(f ) df = 2 Z fc+B 2 fc−B2 N0 2 df = 2× B × N0 2 = B N0 (32)
Modulation
Principle: modulation is all about using of a carrier fc for
transmitting information
Amplitude
s(t) = A(t) cos (2πf (t)t + φ(t))
Phase
Frequency modulation [FM]
Phase modulation [PM]
Amplitude modulation [AM]
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Consequences of modulation
I On analog signals:
frequency band is shifted towards the carrier frequency (⇒ fc)
bandwidth modification
I On digital signals:
power spectral density is shifted
γS(f ) =
1
4[γM(f − fc) + γM(f + fc)] (33) shape of the power spectral density may be modified
Rice’s decomposition
An expression such as
s(t) = A(t) cos(2πfct + φ(t)) (34)
may also be written as
s(t) = A(t) cos(2πfct + φ(t)) (35)
= sI(t) cos(2πfct)−sQ(t) sin(2πfct)
such that
sI(t) = A(t) cos φ(t) (36)
sQ(t) = A(t) sin φ(t) (37)
Rice’s decomposition is essential because, as seen later, it means that any modulated signal can be decomposed into two amplitude modulated signals.
Many receivers in telecommunications use this principle!
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Information theory and channel capacity: there is
maximum bit rate! I
Theorem (Shannon-Hartley)
Channel capacity C (conditions for the error rate Pe → 0)
C [b/s] = B log2 1 + S N (38) where
I B is the channel bandwidth in Hz
I S
On the importance of the
EbN0
ratio for digital transmissions
Assume infinite bandwidth and a Gaussian white channel,C = lim B→ ∞ B log2 1 + S N (39) As
I S = EbRb (Eb is the energy of one bit and Rb = T1b is the
bitrate) I N = B N0 Therefore, C = lim B→ ∞ B log2 1 + EbRb B N0 = lim x→0 log21 + xEbRb N0 x H = log2 e limx→0 ( 1 1 +xEbRb N0 EbRb N0 ) = 1 ln 2 EbRb N0 (40)
At maximum capacity: C = Rb, so that NEb0 = ln 2 ≡ −1.59 [dB] is
the absolute minimum.
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Outline
1 Reminder
2 Representation of bandpass signals
3 Noise in telecommunications systems
4 Digital modulation
5 Spread spectrum
6 Channels for digital communications and intersymbol
interference
7 Navigation systems
8 Multiplexing
9 Telephone traffic engineering
10 Transmission over twisted pairs (fixed telephone network)
Towards a dedicated representation for bandpass signals
Why do we use “frequency-based” representations?
[Systems] it is convenient for linear systems, and most
communication systems are linear (channel, filter, etc).
[Sharing] channels can be shared if signals are “frequency”-friendly (multiplexing).
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What type of
frequency signals
do we have?
1 in its original form, a signal is in its baseband (voice is in
[300, 3400] Hz).
2 bandwidth ≡ band occupied by the signal.
3 for digital signals, sampling −→ sampling frequency (driven by
Nyquist’s criterion and relating to the highest frequency, thus the frequency content).
4 spectrum
1 V(f ) for deterministic signals 2 γV(f ) for stochastic processes.
Purposes of dealing with
bandpass
signals specifically
1 Lower the sampling frequency (not twice the highest
frequency!)
2 There are many more convenient representations of such a
signal (analytic signal, complex envelope, baseband equivalent, quadrature components, etc):
1 They are equivalent, but not equal.
2 Which one is the most appropriate depends on the context (hard to foresee).
3 See these representations as tools!
4 The many representations are convenient theoretical and practical ways to process signals.
5 Almost every receiver uses the underlying theory of bandpass signals.
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Representation of bandpass signals: outline
I Sampling of band-limited signals
I Representations of deterministic band-limited (bandpass) signals:
bandpass equivalent, analytic signal,
complex envelope, and more !
I Representation of bandpass systems
Sampling of bandpass signals
Theorem (“Revised” sampling theorem)
Assume that v (t) is a deterministic, energy limited signal, with a W large bandwidth, whose Fourier V(f ) transform is upper
bounded by fu (that is V(f ) = 0 for f > fu).
Then, it is possible to characterize this signal with samples v [nTs],
n ∈ {−∞, +∞}, taken at a fs sampling frequency if this frequency
is equal to fs = 2fku, where k is the largest integer strictly smaller
than fu
W.
It should be noted that not all sampling frequencies are valid (in order to reconstruct v (t) perfectly), except for all those that are strictly larger than 2fu.
Important practical consequences:
1 sampling frequency: from fs = 2fu to fs = 2fu
k . k times less!
2 there are other ways to look at bandpass signals: new
representations!
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Representations of deterministic bandpass signals
Definition (Bandpass)
A bandpass signal v (t) is a signal if there exist two values B and f0, such that B f0, and
∀f 6∈ f0 − B 2, f0 + B 2 , V(f ) = 0 (41)
Definition (Equivalent baseband)
Assume a deterministic bandpass signal v (t). v (t) is an equivalent
baseband of v (t) if there exists a frequency f0, comprised inside
the frequency band of v (t), such that
v (t) = Rev (t)e2πjf0t+jϕ0 (42)
Note that v in fact covers a family of equivalent baseband signals (since v (t) = (v (t) + jz(t))e−2πjf0t−jϕ0 are all valid candidates).
Representations of deterministic bandpass signals
Definition (Bandpass)
A bandpass signal v (t) is a signal if there exist two values B and f0, such that B f0, and
∀f 6∈ f0 − B 2, f0 + B 2 , V(f ) = 0 (41)
Definition (Equivalent baseband)
Assume a deterministic bandpass signal v (t). v (t) is an equivalent baseband of v (t) if there exists a frequency f0, comprised inside
the frequency band of v (t), such that
v (t) = Rev (t)e2πjf0t+jϕ0 (42)
Note that v in fact covers a family of equivalent baseband signals (since v (t) = (v (t) + jz(t))e−2πjf0t−jϕ0 are all valid candidates).
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Summary of some representations of a bandpass signal
v(t) vI(t) + jvQ(t) va(t) ev(t) ≡ av(t)ejφv(t) ≡ Re(.) ×e−2πjf0t ×e+2πjf0t ⊗ δ(t) + πtj vI(t) cos(.)− vQ(t) sin(.)
Working on the spectrum I
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Working on the spectrum II
Analytic signal I
f
2 1
||H(f)||
Let us consider the following filter that removes the negative frequency components of a signal with the (Heaviside) step function:
H(f ) =
(
0 if f < 0
2 if f ≥ 0 = 1 + sign(f ) (43) whose impulse response is
h(t) = δ(t) + j
πt (44)
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Analytic signal II
Definition (Analytic signal)
A signal which has no negative-frequency components is called an
analytic signal. In the time domain, it is obtained as va(t) = v (t) ⊗ δ(t) + j πt (45) = v (t) + jv (t)⊗ 1 πt (46)
Representation map: analytic signal
v(t) vI(t) + jvQ(t) va(t) ev(t) ≡ av(t)ejφv(t) ≡ Re(.) ×e−2πjf0t ×e+2πjf0t ⊗ δ(t) + πtj vI(t) cos(.)− vQ(t) sin(.) 45 / 495Hilbert transform
Definition (Hilbert transform)
The Hilbert transform of a signal v (t), denoted as v (t), is definede
by
e
v (t) = v (t) ⊗ 1
πt (47)
With this definition,
va(t) = v (t) + jv (t)e (48)
Properties of the Hilbert transform
I The energy (or power) of a signal and that of its Hilbert transform are equal.
I [Hilbert transform of a modulated signal] Assume that v (t) is a baseband signal, then
^
Hilbert transform
Definition (Hilbert transform)
The Hilbert transform of a signal v (t), denoted as v (t), is definede
by
e
v (t) = v (t) ⊗ 1
πt (47)
With this definition,
va(t) = v (t) + jv (t)e (48)
Properties of the Hilbert transform
I The energy (or power) of a signal and that of its Hilbert transform are equal.
I [Hilbert transform of a modulated signal] Assume that v (t) is a baseband signal, then
^ v (t) cos(2πfct) = v (t) sin(2πfct) (49) 47 / 495
Going backwards I
va(t) = v (t) + jv (t)e (50) So, v (t) = Re (va(t)) (51) v(t) vI(t) + jvQ(t) va(t) ev(t) ≡ av(t)ejφv(t) ≡ Re(.) ×e−2πjf0t ×e+2πjf0t ⊗ δ(t) + πtj vI(t) cos(.)− vQ(t) sin(.)Going backwards II
Properties of the analytic signal I it has no negative frequency
I it carries the same power as the original signal
I v (t) can be reconstructed from va(t)
I va(t) is not a real signal. Is that a problem?
Why would we want to use va(t)?
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Baseband representation derived from the analytic signal I
Baseband representation derived from the analytic signal II
Definition (Complex envelope of a signal)
The signal that results from a right-to-left shift of the analytic signal in the frequency domain is named the complex envelope of the signal. It is denoted as ev(t).
Mathematically, the complex envelope and its spectrum are related to the analytic signal as follows:
ev(t) = va(t)e−2πjf0t (53)
Ev(f ) = Va(f + f0) (54)
It remains to find a good practical method to determine the complex envelope!
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Representation map: complex envelope
v(t) vI(t) + jvQ(t) va(t) ev(t) ≡ av(t)ejφv(t) ≡ Re(.) ×e−2πjf0t ×e+2πjf0t ⊗ δ(t) + πtj vI(t) cos(.)− vQ(t) sin(.) Question
Rice components I
ev(t) = vI(t) + jvQ(t) (55) v(t) vI(t) + jvQ(t) va(t) ev(t) ≡ av(t)ejφv(t) ≡ Re(.) ×e−2πjf0t ×e+2πjf0t ⊗ δ(t) + πtj vI(t) cos(.)− vQ(t) sin(.) 53 / 495Theoretical determination of Rice components I
In-phase component: vI(t) vI(t) = Re (ev(t)) (56) = Reva(t)e−2πjf0t (57) = Re(v (t) + jv (t)) ee −2πjf0t (58) = v (t) cos(2πf0t) + v (t) sin(2πfe 0t) (59) Quadrature component: vQ(t) vQ(t) = Im (ev(t)) (60) = Imva(t)e−2πjf0t (61) = −v(t) sin(2πf0t) + v (t) cos(2πfe 0t) (62) Therefore,
Theoretical determination of Rice components II
v (t) = Re (va(t)) (63) = Reev(t)e2πjf0t (64) = Re(vI(t) + jvQ(t))e2πjf0t (65) = vI(t) cos(2πf0t) − vQ(t) sin(2πf0t) (66) v(t) vI(t) + jvQ(t) va(t) ev(t) ≡ av(t)ejφv(t) ≡ Re(.) ×e−2πjf0t ×e+2πjf0t ⊗ δ(t) + πtj vI(t) cos(.)− vQ(t) sin(.) 55 / 495Theoretical determination of Rice components III
cos (2πf
0t)
sin (2πf
0t)
+
π2v
I(t)
v (t)
v
Q(t)
+
−
Theoretical determination of Rice components IV
Any bandpass signal can be seen as the sum of two modulated signals by lowpass signals.
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A practical method to compute Rice components I
Theoretically,
+
π2v (t)
cos (2πf
0t)
Hilbert
sin (2πf
0t)
+
+
v
I(t)
A practical method to compute Rice components II
v (t)× 2 cos(2πf0t) = 2 [vI(t) cos(2πf0t)− vQ(t) sin(2πf0t)] cos(2πf0t) = 2vI(t) cos2(2πf0t)− vQ(t) sin(2πf0t) cos(2πf0t)
= vI(t) + vI(t) cos(4πf0t)− vQ(t) sin(4πf0t) (67)
And after a low-pass filter,
v (t)× 2 cos(2πf0t) −→ lowpass filter −→ vI(t) (68)
Likewise,
v (t)× 2 sin(2πf0t) −→ lowpass filter −→ −vQ(t) (69)
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A practical method to compute Rice components III
+
π2v
I(t)
2 sin (2πf
0t)
2 cos (2πf
0t)
−v
Q(t)
v (t)
Amplitude/phase representation of the complex envelope I
ev(t) = vI(t) + jvQ(t) = av(t) ejφv(t) (70) v(t) vI(t) + jvQ(t) va(t) ev(t) ≡ av(t)ejφv(t) ≡ Re(.) ×e−2πjf0t ×e+2πjf0t ⊗ δ(t) + πtj vI(t) cos(.)− vQ(t) sin(.) 61 / 495Amplitude/phase representation of the complex envelope II
By substituting ev(t) by its amplitude + phase description, we have
v (t) = Re ev(t)e2πjf0t (71) = Reav(t) ejφv(t)e2πjf0t (72) = av(t) cos (2πf0t + φv(t)) (73) with, by definition, av(t) = q vI2(t) + vQ2(t) (74) φv(t) = tan−1 vQ(t) vI(t) (75)
Amplitude/phase representation of the complex envelope
III
Conclusions for v (t) = av(t) cos (2πf0t + φv(t))
1 the amplitude av(t) of the complex envelope is the envelope
of the original signal v (t).
2 the instantaneous phase of v (t) is given by the phase of the
complex envelope.
Any bandpass signal can be seen as a signal modulated both in amplitude and in phase.
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Linear bandpass systems
Assume a bandpass filter, around f0, whose response filter is
h(t) = Reeh(t)e2πjf0t (76)
The filtered signal is given by
y (t) = v (t)⊗ h(t) (77) = Z +∞ −∞ h(λ)v (t − λ)dλ (78) Thesis: ey(t) = 1 2 eh(t)⊗ ev(t) (79)
Linear bandpass systems
Assume a bandpass filter, around f0, whose response filter is
h(t) = Reeh(t)e2πjf0t
(76) The filtered signal is given by
y (t) = v (t)⊗ h(t) (77) = Z +∞ −∞ h(λ)v (t − λ)dλ (78) Thesis: ey(t) = 1 2 eh(t)⊗ ev(t) (79) 65 / 495
Proof I
y (t) = v (t)⊗ h(t) = Z +∞ −∞ h(λ)v (t − λ)dλ (80) To get rid of Re (), we useRe (a + jb) = a + jb 2 + a− jb 2 = a + jb 2 + a + jb 2 ∗ (81) So, v (t − λ) = 12 ev(t − λ)e2πjf0te−2πjf0λ + ev∗(t − λ)e−2πjf0te2πjf0λ
and h(λ) = 12 eh(λ)e2πjf0λ + eh∗(λ)e−2πjf0λ Therefore, y (t) = 1 4e 2πjf0t Z +∞ −∞ eh(λ)ev(t − λ)dλ + 1 4e −2πjf0t Z +∞ −∞ eh∗(λ)ev∗(t − λ)dλ + 1 4e −2πjf0t Z +∞ −∞ eh(λ)ev∗(t − λ)e4πjf0λdλ + 1 4e 2πjf0t Z +∞ −∞ eh∗(λ)ev(t − λ)e−4πjf0λdλ
Proof II
y (t) = 1 2Re Z +∞ −∞ eh(λ)ev(t − λ)dλe2πjf0t (82) + 1 2Re R+∞ −∞ eh∗(λ)ev(t − λ)e−4πjf0λdλe2πjf0t (83)But, eh∗(λ)ev(t − λ) is low frequency and e−4πjf0λ is high
frequency. Therefore, Z +∞ −∞ e ∗ h(λ)ev(t − λ)e−4πjf0λdλ ≈ 0 (84) Finally, y (t) = 1 2Re ((eh(t) ⊗ ev(t)) e 2πjf0t) (85) or ey(t) = 1 2 eh(t)⊗ ev(t) (86) 67 / 495
Bandpass filtering
Use of ey(t) = 12 eh(t)⊗ ev(t)I the output is a bandpass signal too (⇒ it has a baseband equivalent).
I we have a way to filter a bandpass signal by its baseband equivalent. This is really helpful when we work with digital signals (because the sampling frequency is much lower).
Analytic signal of a stochastic process
Two signals can be considered:
1 the stochastic process: X (t)
2 its auto-correlation function ΓXX (τ ) or its power spectrum
γX(f )
The theory of equivalent baseband signals applies both to X (t) and to ΓXX (τ ), but they are very different concepts.
Remember that when a stochastic process X (t) passes through a (linear) filter, its power spectrum is multiplied by kH(f )k2.
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Link between a stochastic process and its complex envelope
X (t) is related to its complex envelope by X (t) = ReeX(t) e2πjf0t
(87) X (t) being a stochastic process, its complex envelope eX(t) is also
a stochastic process.
In general, X (t) is not stationary because its mean is time dependent.
Solution: introduction of a random phase Θ uniformly distributed over [0, 2π[
X (t) = ReeX(t) ej(2πf0t+Θ)
Rice decomposition of a stochastic process I
Likewise to developments for deterministic signals, the in-phase and quadrature components of a stochastic process can be expressed as eX(t) = XI(t) + j XQ(t) (89)
Rice components of the X (t) stochastic process are then obtained by X (t) = ReeX(t) e2πjf0t (90) = Re(XI(t) + j XQ(t)) e2πjf0t (91) = XI(t) cos(2πf0t)− XQ(t) sin(2πf0t) (92)
By analogy with the developments for deterministic signals, we build the analytic signal by filtering X (t) with the following filter H(f ) that removes all components for negative frequencies:
H(f ) =
(
0 if f < 0
2 if f ≥ 0 (93)
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Rice decomposition of a stochastic process II
By applying Wiener-Kintchine, the power spectral density of the analytic signal is given by
γXa(f ) = kH(f )k2γX(f ) (94) =
(
4 γX(f ) if f ≥ 0
0 if f < 0 (95) Note that it can also be written as γXa(f ) = 2H(f ) γX(f ).
Power spectrum
γeX(f ) = γXa(f + f0) (96) ΓXX (τ ) = 1 2Re (ΓXaXa (τ )) (97) = 1 2Re ΓeXeX (τ ) e 2πjf0τ (98)It is shown in a later chapter that, after stationarization, we get
γX(f ) = γeX(f − f0) + γ
∗
eX(−f − f0)
4 (99)
Important practical result: γX(f ) can be derived from γeX().
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Outline
1 Reminder
2 Representation of bandpass signals
3 Noise in telecommunications systems
4 Digital modulation
5 Spread spectrum
6 Channels for digital communications and intersymbol
interference
7 Navigation systems
8 Multiplexing
9 Telephone traffic engineering
10 Transmission over twisted pairs (fixed telephone network)
Steps
1 Understand the origin of noise (thermal noise)
2 Look for a model of noise (in terms of the power spectral
density of a stochastic voltage)
3 Find a corresponding model for the noise getting out of a
one-port circuit (12kBT )
4 Define a practical way to derive the amount of noise when
dealing with a two-port circuit (noise figure, F0)
5 Find formulas to calculate the amount of noise accumulated
by the different elements of a cascade of two-port circuits (F0
for the cascade)
6 Derive good engineering rules for the order of elements in a
cascade
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Noise in telecommunications systems I
Noise is a recurrent issue/problem in telecommunications systems. Remember for example the following theorem:
Theorem (Shannon-Hartley)
Channel capacity C (conditions for the error rate Pe → 0)
C b s = B log2 1 + S N (100) where
I B is the channel bandwidth in Hz
I S
N the signal-to-noise ratio (in watts/watts, not in dB).
Noise in telecommunications systems II
There are several physical sources of noise:
I thermal noise
I shot noise
I ...
There are also “system” sources of noise
I quantization noise I intermodulation noise I crosstalk I interference noise I ... 77 / 495
Noise in telecommunications systems III
Approach to deal with noise in telecommunications systems?
1 model the most common noise: thermal noise for electronic
circuits
2 the principles of thermal noise can also be used to model
Modeling noise in electronic circuits
1 Characterization of a one-port circuit (source) Available noise power
Noise temperature of a one-port circuit Signal to noise ratio
2 Characterization of a two-port circuit (channel, amplifiers,
filters, etc)
Gain
Noise factor Figure of merit
Effective noise temperature Particular case: attenuator
3 Cascade of two-port circuits (a complete chain)
79 / 495
Thermal noise: a measuring experiment
R E (t)
(a) (b)
I (t) = E (t)R noisy R
Figure: (a) Physical circuit with noise, (b) Th´evenin equivalent circuit for a resistor considered as a noise generator.
A conductive element with two terminals (≡ one-port circuit) may be characterized by:
I its resistance R.
I Free electrons have some random motion depending on the temperature T ⇒ noise voltage source E (t).
Thermal noise: towards a model for E(t) I
A natural source of noise is thermal noise, caused by the omnipresent
motion of free electrons in conducting material.
I Because a voltage that would be measured at the output of a
resistance R is produced by many free electrons, by the central limit theorem, the amplitude statistics are modeled by a Gaussian.
The thermal noise has a zero mean (can be shown analytically and experimentally).
It can be shown that the autocorrelation function of thermal noise is well modeled by
ΓEE (τ ) = kBTRe −|τ|/t0 t0 (101) where kB = 1.38× 10−23 [J/K] is Boltzmann’s constant.
T = 273.15 + C is the absolute temperature of the resistor (in Kelvin); C is the temperature in Celsius.
t0 = 10−12[s] is the average time between collisions of electrons.
81 / 495
Thermal noise: towards a model for E(t) II
As the autocorrelation function is
ΓEE (τ ) = kBTRe −|τ|/t0 t0 (102) the PSD is γE(f ) = 2kBTR 1 + (2πft0)2 (103)
Empirically, at room temperature, for low frequencies
(< 1000 [GHz]), the noise PSD is almost flat so that we may take
γE(f ) ' 2kBTR (104)
γE(f ) corresponds to the power of E (remember that the power is in the form of E2(t)).
Thermal noise: towards a model for E(t) III
Note: for a B bandwidth, a resistor in a short circuit dissipates a noise power of (non sinusoidal signal):
P = Z ∞ −∞γE(f )df /R = 2 Z ∞ 0 γE(f )df /R = 4kBT B (105)
Noise power spectral density
Consider a thermal noise with a noise temperature of T = 290 [K] and R = 1 [Ω], then we have
γE(f ) = 2 × 1.38 × 10−23 × 290 × 1 = 8 × 10−21 W Hz (106) Note that γE(f ) is the power due to thermal noise; it is not the
power available at the output!
In the following, we calculate the available noise power.
83 / 495
Characterization of a single-port circuit (dipole)
Source impedance: Zs (f ) = Rs (f ) + jXs (f ) (107) Load impedance: ZL(f ) (108)Z
sZ
LE
Available power
Definition (Available power)
The available power is the maximum power that can be drawn from a source.
Theorem (Maximum power transfer)
The maximum power transfer occurs when the load impedance is equal to the conjugate of the source impedance (matched
impedances):
ZL(f ) = ZS∗(f ) (109)
If impedances are almost purely resistive, then
RL = RS (110)
85 / 495
The case of sinusoidal signals I
When the signals are sinusoidal, the power provided by a source S at the output of the dipole, PpS, is given by
PpS = lim T→∞ 1 T Z T 0 v (t)i(t)dt (111) = 1 2Re b VbI∗ (112)
where V andb bI are phasors are defined with peak values, instead of root mean square (rms) values, and therefore there is a 12 factor.
The case of sinusoidal signals II
Zs ZL E I VIn a load ZL, we have (voltage divider): b V = ZL Zs + ZL b E and bI = Eb Zs + ZL Therefore, PpS = 1 2Re b VbI∗ = 1 2Re ZLEb Zs + ZL b E∗ (Zs + ZL)∗ ! = Re (ZL) bE 2 2kZs + ZLk2 (113) 87 / 495
The case of sinusoidal signals III
So, the available power (that is when ZL(f ) = ZS∗(f ), so that
Zs + ZL = 2 Re (Zs)) from the Source is PaS = Re (ZL) bE2 2kZs + ZLk2 = Re (ZL) bE 2 8 Re (Zs)2 = Eb 2 8 Re (Zs) (114)
The case of sinusoidal signals IV
Summary
The available power in the load is PaS =
b E2 8 Re (Zs)
(115) By definition, the effective power produced by the source is (for an open circuit, matched impedance, so that Zs + ZL = 2 × Re (Zs))
PS = 1 2EbIb ∗ = 1 2Eb b E∗ (Zs + ZL)∗ = Eb 2 4 Re (Zs) (116)
It follows then that the power delivered by the Th´evenin generator and the power dissipated in the generator’s Th´evenin resistance are the same.
89 / 495
Available noise power (matched impedance)
For an arbitrary random noise source, the provided noise power is PpN = lim T→+∞ 1 T Z T 0 V (t)I (t)dt (117) R E R I V
For the particular case of thermal noise
The available power spectral density is, by applying Wiener-Kintchine: γaN(f ) = kH(f )k2γE(f ) = R R + R 2 2kBT = kBT 2 (118)
Remember that this assumes the adaptation of the impedance (Zs = ZL∗); otherwise, the result would depend on the impedances!
Noise at the output of an antenna pointing towards the sky
Sky temperature R γE(f ) = 2kBTa Ta Noiseless resistor Noise source Example Contributions to Ta:I the sky contributes to 10 [K] (3 [K] of residual temperature of the Big Bang + 7 [K] due to atmospheric absorption).
I the ground contribution is typically 0.1 of 290 [K] (contributions of secondary lobes looking at the ground)
Thus,
Ta = 0.9× 10 +0.1× 290 = 38 [K] (119)
The noise PSD is then
γaN(f ) = kBT 2 = 2.6 × 10 −22 hW Hz i (120) 91 / 495
A note on matched impedances
There exist two ways to match loads:
1 Z∗
S = ZL (conjugate matching). This ensures the maximum
transfer of power (that is the “available power”).
2 Zc = ZL, where Zc denotes the characteristic impedance of a
transmission line. Zc is the ratio of the amplitudes of voltage
and current of a single wave propagating along the line. When Zc = ZL, there is no reflection.
In practice, what should we do when we connect a circuit to a line?
I ZS∗ = ZL is mandatory. It is the matching condition for this
chapter.
I Luckily, Zc = Rc + jLc and Lc Rc, so that Zc ' Rc.
Summary I
Arbitrary load Matched load
Sinusoidal signals PpS = 12Re b VbI∗ = bE2Re(ZL) 2kZs+ZLk2 PaS = bE2 8Re(Zs)
Stochastic processes PpN = limT→+∞ T1 R0T V (t)I (t)dt γaN(f ) = γE(f ) 4Re(Zs)
Thermal noise γaN(f ) = kB2T
Table: Power provided by a one-port circuit.
Theorem (Available power)
The available power from a thermal source in a bandwidth of B is
PN = Z +∞ −∞ γaN(f ) df = 2× Z f0+B 2 f0−B2 kBT 2 df = kBTB (121) Conclusions: 93 / 495
Summary II
We have a model for the power of (thermal) noise
PN = kBTB (122)
It does not depend on the impedance (because we assume impedance matching). It depends only on
I the temperature T
I the bandwidth B
Noise power
Consider a thermal noise with a noise temperature of T = 290 [K] and B = 10 [MHz], then we have
PN(f ) = 1.38× 10−23 × 290 × 107 = 40 × 10−15 [W] (123)
Summary III
PN = kBTB (124)
So the noise power increases with the bandwidth ⇒ we may want to reduce the bandwidth
However, the capacity is given by C = B log2 1 + S N (125) If S is fixed and we only have a white noise, then
C = B log2 1 + S kBTB (126) To increase the capacity C , there is no alternative other than to increase the bandwidth ⇒ trade-off !
95 / 495
Noise temperature for a one-port circuit [generic model]
Definition (Noise temperature [at a given frequency])
The noise temperature at a given frequency is the absolute temperature that an impedance should have to produce, by
thermal effect, for a given frequency, a noise power spectral density equal to that of the circuit.
By definition thus,
γaN(f ) = kBT (f )
2 (127)
Definition (Frequency average for T (f ) and bandwidth)
The maximum temperature, denoted T, the noise temperature of the dipole and the bandwidth are defined such that
Noise temperature for a one-port circuit [generic model]
Definition (Noise temperature [at a given frequency])
The noise temperature at a given frequency is the absolute temperature that an impedance should have to produce, by
thermal effect, for a given frequency, a noise power spectral density equal to that of the circuit.
By definition thus,
γaN(f ) = kBT (f )
2 (127)
Definition (Frequency average for T (f ) and bandwidth)
The maximum temperature, denoted T, the noise temperature of the dipole and the bandwidth are defined such that
PaN = kBTB (128)
97 / 495
Signal to noise ratio of a one-port circuit
Definition
The Signal to Noise ratio (S/N) of a dipole is defined as the ratio between the available power of the signal and the noise power
S
N =
PaS
PaN (129)
Warning: in the following S denotes the Signal and not the noise source!
By convention, if the signal is modulated, the power of the useful signal is defined as follows (it is just a reference to compare
techniques):
I for amplitude or angular modulations, we consider the power of the carrier (it is then a carrier to noise ratio NC),
I for suppressed carrier amplitude modulation techniques, we take the mean power of the modulating signal, and
Characterization of a two-port circuit (quadripole) I
1
1
2
2
linear
quadripole
output
Z
SE
V
input
Figure: Scheme of a two-port circuit.
99 / 495
Characterization of a two-port circuit (quadripole) II
Steps:
I Notion of gain?
I Characterization of the amount of internal noise of the
quadripole by means of the normalized notion of noise figure
I Equivalent circuits
I Figure of merit (not normalized)
I Effective noise temperature
I Special case: purely resistive attenuator
Hypothesis: loads are matched (complex conjugate) at the input and at the output
Noise figure of a two-port circuit
Definition (Noise Figure (NF), F0)
Assuming fixed internal impedances, the spot noise figure of a two-port circuit, for a given frequency f , denoted F0(f ), is the
ratio between
(1) the noise power spectral density at the output of the quadripole, for the appropriate frequency f , when the noise temperature of the generating one-port put at the input is normalized to be T0 = 290 [K ], and
(2) the contribution of the generating input source, at a frequency f , to the noise power spectral density at the output.
1 1 2 2 quadripole Rs Rin= Rs γaN1(f ) = 1 2kBT0 γaN2(f ) = G (f )γaN1(f ) + γaNq(f ) linear 4kBTRsB 101 / 495
Property: F
0>
1
1 1 2 2 quadripole Rs Rin = Rs γaN1(f ) = 1 2kBT0 γaN2(f ) = G (f )γaN1(f ) + γaNq(f ) linear 4kBTRsB So, by definition, F0(f ) = γaN2(f ) G (f )γaN1(f ) = G (f )γaN1(f ) + γaNq(f ) G (f )γaN1(f ) > 1 (130)Equivalent circuit
F0(f ) = G (f )γaN1(f ) +γaNq(f ) G (f )γaN1(f ) ⇐⇒ γaNq(f ) = [(F0(f )− 1) γaN1(f )] G (f ) G G γaN2(f ) = G (f )γaN1(f ) + γaNq(f ) γaN1(f ) γaN1(f ) γaN2(f ) = G (f )F0(f )γaN1(f ) (F0(f )− 1)γaN1(f ) γaNq(f )Figure: Scheme of a noisy quadripole and an equivalent circuit (we model the internal noise by (F0(f )− 1) γaN1(f ) at the entrance)
103 / 495
Interpretation of the notion of noise figure
The input signal to noise ratio is given by
S N in = γin(f ) γaN1(f ) (131) At the output of the two-port circuit, we have
S N out = γout(f ) γaN2(f ) (132) Therefore, S N in S N out = γin(f ) γaN1(f ) γaN2(f ) γout(f ) (133)
As γout(f ) = G(f )γin(f ) , this ratio becomes
S N in S N out = γaN2(f ) γaN1(f )G (f ) = G (f )F0(f )γaN1(f ) γaN1(f )G (f ) = F0(f ) (134)
Noise figures for un-normalized temperatures (
T
s6= T
0)
What if the input temperature Ts 6= T0?
This leads to define a different notion ⇒ the figure of merit F. Link between F and F0 (F0 is provided by the manufacturer)?
Remember that, considering γaN1(f )|T =T0 =
1
2kBT0:
γaNq(f ) = (F0 − 1) 1
2kBT0G (f ) (135) The internal noise of a two-port circuit is independent of the input temperature (the last is just a convention). Therefore, another temperature Ts then leads to another figure of merit F . It is
derived as follows γaNq(f ) = (F0 − 1) 1 2kBT0G (f ) = (F − 1) 1 2kBTsG (f ) (136) and, finally, F = 1 + T0 Ts (F0 − 1) (137) 105 / 495
Effective noise temperature of a two-port circuit I
G G γaN2(f ) = G (f )γaN1(f ) + γaNq(f ) γaN1(f ) γaN1(f ) γaN2(f ) = G (f )F0(f )γaN1(f ) (F0(f )− 1)γaN1(f ) γaNq(f ) γaN2(f ) = 1 2kBT0G (f ) + γaNq(f ) = 1 2kB [T0 +(F0 − 1)T0] G (f )
Effective noise temperature of a two-port circuit II
Definition (Effective (input-)noise temperature)
Te = (F0 − 1)T0 (138)
It is the additional temperature required for an input source to produce the same available power at the output.
Note that:
Te = (F0 − 1)T0 ⇔ F0 = 1 + Te
T0
(139)
Noisy two-port circuit
Consider an effective noise temperature Te = 120 [K], then
F0 = 1 + 120
290 = 1.41 = 1.5 [dB] (140)
107 / 495
Attenuator with a “Gain” G = 1/L or Loss L
Let us take an attenuator (for example a purely resistive circuit or a lossy transmission line) at temperature T0.
(1) Assuming matched input and output impedances, the available noise power at the output is
γaN2(f ) = 1
2kBT0 (141)
(2) But that the attenuator is characterized by its effective noise temperature Te, then the output noise power is
γaN2(f ) = 1
2kB (T0 + Te) 1
L (142)
By combining these two expressions: Te = (L − 1)T0. So that,
F0 = 1 + (L− 1)T0 T0
= L (143)
Conclusion: attenuators have a noise figure F0 equal to their
Attenuator with a “Gain” G = 1/L or Loss L
Let us take an attenuator (for example a purely resistive circuit or a lossy transmission line) at temperature T0.
(1) Assuming matched input and output impedances, the available noise power at the output is
γaN2(f ) = 1
2kBT0 (141)
(2) But that the attenuator is characterized by its effective noise temperature Te, then the output noise power is
γaN2(f ) = 1
2kB (T0 + Te) 1
L (142)
By combining these two expressions: Te = (L − 1)T0. So that,
F0 = 1 +
(L− 1)T0
T0 = L (143)
Conclusion: attenuators have a noise figure F0 equal to their
attenuation ratio L when their physical temperature equals T0.
109 / 495
What about the attenuator L at other temperatures?
Let us consider Ts 6= T0 and calculate F .
There are two possible ways to calculate F :
1 same reasoning as previously: it is impossible to discriminate
the output of the two-port circuit from the input one-port circuit, so that γaN2(f ) = 1 2kBTS = γaN2(f ) = 1 2kB (TS + Te) 1 L (144) and F = L.
2 Remember that F0 represents the signal to noise degradation
F0 = S N in/ S N out = L (145) But, as L =Sin/Sout, we have Nin = Nout.
Conclusion: for an attenuator with a factor L, the amount of noise is always unaffected, so that
Noise figure of two-port networks I
γaN1(f ) (F01− 1)γaN1(f ) (F02− 1)γaN1(f ) G1F01γaN1(f ) G2 G1 F02 F01Figure: Cascading two-port elements.
For a two-port network with 2 stages, F0 = γaN1(f )G1G2 + (F01 − 1) γaN1(f )G1G2 + (F02 − 1) γaN1(f )G2 γaN1(f )G1G2 = 1 + (F01 − 1) + (F02 − 1) G1 (147) = F01 + (F02 − 1) G1 (148) 111 / 495
Noise figure of two-port networks II
For a two-port network with n stages, F0 = F01 + F02 − 1 G1 + F03 − 1 G1G2 +· · · = F01 + n X i=2 F0i − 1 Qi−1 j=1Gj (149) Likewise, Te = Te1 + Te2 G1 + Te3 G1G2 +· · · = Te1 + n X i=2 Tei Qi−1 j=1Gj (150)
Low-noise amplifier for receivers
F0 = F01 + F02 − 1 G1 + F03 − 1 G1G2 + · · · (151) Two consequences:1 the noise figure always increases with an additional stage. 2 the overall noise figure of a receiver is primarily set by the
noise figure of its first amplifying stage.
Therefore, the first stage amplifier is often a Low-Noise Amplifier (LNA). Then, the overall receiver noise figure is
Freceiver ' FLNA + Fothers − 1GLNA (152)
113 / 495
Outline
1 Reminder
2 Representation of bandpass signals
3 Noise in telecommunications systems
4 Digital modulation
5 Spread spectrum
6 Channels for digital communications and intersymbol
interference
7 Navigation systems
8 Multiplexing
9 Telephone traffic engineering
10 Transmission over twisted pairs (fixed telephone network)
Digital modulation
1 Criteria ruling the selection of a modulation scheme 2 Definition and typology of digital modulations
3 Classic linear modulations Description
Determination of the power spectrum Amplitude modulation (ASK)
Phase modulation (PSK)
Quadrature modulation (QPSK) 4 Offset modulations
Description
Determination of the power spectrum Offset quadrature modulation (OQPSK)
Minimum shift modulation techniques (MSK)
115 / 495
Criteria for choosing of a modulation scheme
1 Resistance to distortion and perturbation. That includes: resistance to additive white Gaussian noise. Noise usually results in a bit error rate Pe, expressed in terms of the Eb/N0
ratio.
sensitivity to interference (multipath, other users, etc). sensitivity to imperfect filters. This is associated to the phenomenon of intersymbol interference.
sensitivity to non-linearities. 2 Spectral occupancy:
1 spectral efficiency η, expressed in bit per second per Hertz
h
b/s Hz
i
, which measures the bit rate that can be transmitted per unit of frequency bandwidth for a given modulation.
2 asymptotic behavior, defined by the values of the spectral density for frequencies relatively distant from the carrier frequency.
3 Simplicity of implementation.
117 / 495
Spectral efficiency of linear modulation techniques
C = B log21 + EbRb B N0 max: R b→C =⇒ η ' Rb B = log2 1 +ηEb N0
Understanding digital modulations
We will make an intensive use of alternative representations
v(t) vI(t) + jvQ(t) va(t) ev(t) ≡ av(t)ejφv(t) ≡ Re(.) ×e−2πjf0t ×e+2πjf0t ⊗ δ(t) + πtj vI(t) cos(.)− vQ(t) sin(.) 119 / 495
Reminder: building a digital signal
P+∞k=−∞A
kp(t
− kT )
−1 −1
+1
Information: Pulse shape:
pulse shapes Combining modulated Result: A0 A1 A2 Ak p(t) t t t p(t− kT ) p(t− T ) p(t− 2T ) −T /2 T /2 T t t t t A1p(t− T ) A0p(t) A2p(t− 2T ) +X∞ k=−∞ Akp(t− kT )
Main characteristics of digital signals:
1 With digital signals, the fundamental unit is a time slot T . 2 One information symbol Ak per time slot T (no overlap).
p(t − kT ) thus also acts as a time window!
3 Same pulse shape for each time slot (eases the task of the
Reminder: building a digital signal
P+∞k=−∞A
kp(t
− kT )
−1 −1
+1
Information: Pulse shape:
pulse shapes Combining modulated Result: A0 A1 A2 Ak p(t) t t t p(t− kT ) p(t− T ) p(t− 2T ) −T /2 T /2 T t t t t A1p(t− T ) A0p(t) A2p(t− 2T ) +X∞ k=−∞ Akp(t− kT )
Main characteristics of digital signals:
1 With digital signals, the fundamental unit is a time slot T . 2 One information symbol Ak per time slot T (no overlap).
p(t − kT ) thus also acts as a time window!
3 Same pulse shape for each time slot (eases the task of the
receiver).
121 / 495
Definition and typology of digital modulations I
Definition (General expression for digital modulations (based on the complex envelope))
s(t) = Reψ [m(t)]ej(2πfct+ϕc) (153)
The complex function ψ [m(t)], which is related to the modulating waveform m(t), defines the type of modulation. It is also the
complex envelope es(t) of the modulated signal s(t).
Depending on the form of ψ(.) = ψI(.) + j ψQ(.), we generally
distinguish:
I linear modulations for which ψ [m(t)] is a linear function of m(t).
I angular modulations for which ψ [m(t)] has the form of
ψ [m(t)] = ejϕ[m(t)] (154) where ϕ [m(t)] is a linear function of m(t).
Definition and typology of digital modulations II
The modulated signal can also be expressed as
s(t) = ψI [m(t)] cos (2πfct + ϕc)− ψQ [m(t)] sin (2πfct + ϕc)
(155) and
s(t) = kψ [m(t)]k cos (2πfct + ϕc + arg ψ [m(t)]) (156)
123 / 495
Definition and typology of digital modulations III
In the following, we will concentrate on modulations that can be written as s(t) = Re +X∞ k=−∞ dk(t) ej(θk−2πfckT ) ej(2πfct+ϕc) (157)
Two types of linear modulations will be studied:
1 “classic”modulations, for which θk = 2πfckT 2 offset modulations, for which θk = 2πfckT + kπ
2
Classic linear modulations
Description
Classic linear modulations are such that θk = 2πfckT. Therefore,
we have that
s(t) = Rees(t) ej(2πfct+ϕc)
(158) The complex envelope takes the form
es(t) = +∞ X k=−∞ dk(t) (159) = +∞ X k=−∞ Dk pk(t − kT ) (160)
where Dk=Ak+jBk is complex and Ak, Bk are two real random
variables.
In most cases, the pulse shape of pk(t − kT ) is the same for each
symbol k. Therefore, pk(.) becomes p().
125 / 495
Complex envelope of classic linear modulation techniques
The complex envelope can also be written as es(t) = sI(t) + jsQ(t), so that sI(t) = +∞ X k=−∞ Ak p(t − kT ) (161) sQ(t) = +∞ X k=−∞ Bk p(t − kT ) (162) resulting in s(t) = sI(t) cos (2πfct + ϕc)− sQ(t) sin (2πfct + ϕc) (163)
and, by replacing sI and sQ by their value,
s(t) = " +∞ X k=−∞ Ak p(t− kT ) # cos (2πfct + ϕc)− " +∞ X k=−∞ Bk p(t− kT ) # sin (2πfct + ϕc)
Derivation of the power spectral density I
Power spectral density of a modulated signal?
The modulated signal is a stochastic process S(t) that can be written, taking ϕc = 0, as
S(t) = ReM(t) ej2πfct (164)
where M(t) is a complex stochastic process (such as the complex envelope es(t) in our case).
But is S(t) stationary?
Obviously, we take M(t) stationary. However, even then
µS = E {S(t)} = µM ej2πfct (165)
is time-dependent (not constant).
Therefore, as such, we cannot calculate γS(f ).
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Derivation of the power spectral density II
Derivation of the power spectral density III
Stationarization
The S(t) process is not stationary because its mean is time-dependent. We have to stationarize the signal.
For that purpose, we add the random phase Θ whose probability density function (pdf) is uniformly distributed over [0, 2π[ (in other words, pdfΘ(θ) = 2π1 for θ ∈ [0, 2π[ and 0 outside):
S(t) = ReM(t) ej(2πfct+Θ) (166)
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Derivation of the power spectral density IV
Mean of S(t) = ReM(t) ej(2πfct+Θ)?
Note that M(t) and Θ are independent. For the computation, we do
1 Re (a + jb) is replaced by Re (a + jb) = a+jb
2 +
a−jb 2
2 Then, we take the expectation of both terms: the expectation
of a sum is the sum of the expectations.
3 Then, for the first term, we have
1 2E n M(t) ej(2πfct+Θ)o = 1 2E {M(t)} E n ej(2πfct+Θ)o = 1 2µM Z 2π 0 e j(2πfct+Θ) 1 2πdθ = 1 2µM × 0 = 0 µS = E {S(t)} = 0 (167)
Derivation of the power spectral density V
Autocorrelation function of S(t)? ΓSS (t, t − τ) = E {S(t) S(t − τ)} (168) As S(t) = ReM(t) ej(2πfct+Θ) (169) = 1 2 h M(t) ej(2πfct+Θ) + M∗(t) e−j(2πfct+Θ)i (170) we have S(t)S(t − τ) = 1 2 h M(t) ej(2πfct+Θ) + M∗(t) e−j(2πfct+Θ)i (171) × 1 2 h M(t − τ) ej(2πfc(t−τ)+Θ) + M∗(t − τ) e−j(2πfc(t−τ)+Θ)i 131 / 495Derivation of the power spectral density VI
S(t)S(t − τ) = 1 4M(t) e j(2πfct+Θ)M(t − τ) ej(2πfc(t−τ)+Θ) (172) + 1 4M(t) e j(2πfct+Θ)M∗(t − τ) e−j(2πfc(t−τ)+Θ)(173) + 1 4M ∗(t) e−j(2πfct+Θ)M(t − τ) ej(2πfc(t−τ)+Θ)(174) + 1 4M ∗(t) e−j(2πfct+Θ)M∗(t − τ) e−j(2πfc(t−τ)+Θ) S(t)S(t − τ) = 1 4M(t)M(t − τ)e j(2πfc(2t−τ)+2Θ) (175) + 1 4M(t)M ∗(t − τ)ej2πfcτ (176) + 1 4M ∗(t)M(t − τ)e−j2πfcτ (177) + 1 4M ∗(t)M∗(t − τ)e−j(2πfc(2t−τ)+2Θ) (178)
Derivation of the power spectral density VII
Because, ΓSS (t, t − τ) = E {S(t)S(t − τ)}, the terms of E {.}
containing 2Θ are null. Therefore, we are left with
ΓSS (t, t − τ) = 1 4E M(t) M∗(t − τ) ej2πfcτ + M∗(t) M(t − τ) e−j2πfcτ = 1 4E 2 Re M(t) M∗(t − τ) ej2πfcτ (179) = 1 2Re E M(t) M∗(t − τ) ej2πfcτ (180) = 1 2Re ΓMM (t, t − τ) e j2πfcτ (181) Finally, because ΓSS (τ ) = 1 4 ΓMM (τ ) ej2πfcτ + ΓMM(τ )∗ e−j2πfcτ (182)
and x∗(t) ↔ X∗(−f ), we have that
γS(f ) = γM (f − fc) + γ ∗
M (−f − fc)
4 (183)
where γM(f ) is the power spectral density of M(t).
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Derivation of the power spectral density VIII
Power spectral density of a modulation signal for classic linear modulation techniques If M(t) is real: γS(f ) = γM (f − fc) + γM (f + fc) 4 (184) If M(t) is complex: γS(f ) = γM (f − fc) + γM∗ (−f − fc) 4 (185)
Calculation of the power spectrum of the complex envelope
(for classic linear modulation techniques)
The complex envelope of the modulated signal is (es(t) = M(t))
M(t) = +∞
X
k=−∞
Dk p(t − kT ) (186)
The sequence of complex random variables Dk is characterized by
I mean: µD = E {Dk}
I variance: σD2 = E {(Dk − µD) (Dk − µD)∗}
I autocorrelation function: ΓDD (k, k − l) = E {DkDk∗−l}
I covariance function: CDD(k, k − l) = E {(Dk − µD) (Dk−l − µD)∗}
After the stationarization of Dk, the PSD of the baseband complex
envelope is (cf. first course in telecommunications)
γM(f )= kP(f )k 2 T " σ2D +kµDk2 +∞ X m=−∞ 1 Tδ f − m T # (187)
In conclusion (γM(f ) being real in this case):
γS(f ) = γM (f − fc) + γM(f + fc)
4 (188)
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Amplitude Shift Keying (amplitude modulation) I
Definition (Complex envelope of the Amplitude Shift Keying (ASK))
es(t) =
+X∞ k=−∞
Ak p(t − kT ) (189)
Common choice for the shaping pulse function over [0, T ]:
p(t) = rect[0,T ](t) (190)
Goal of the following slides: we need to find the envelope a(t) and the phase ϕ(t) of the modulated signal. These two signals can
be derived from
Amplitude Shift Keying (amplitude modulation) II
Trick: ±Ak can be expressed as ±Ak = kAkk e(1−sign(Ak))
π
2j
Let us verify this:
I Ak is positive (Ak = kAkk): Ak = kAkk e(1−sign(Ak)) π 2j =kAkk e(1−1)π2j = kAkk e0π2j = kAkk (192) I Ak is negative (Ak =−kAkk): Ak =kAkk e(1−(−1)) π 2j = kAkk e2π2j =kAkk eπj =−kAkk (193) So, we have es(t) = +∞ X k=−∞ Ak p(t − kT ) (194) = +∞ X k=−∞ kAkk e(1−sign(Ak)) π 2j p(t − kT ) (195) 137 / 495
Amplitude Shift Keying (amplitude modulation) III
Therefore, the envelope a(t) and the phase ϕ(t) of the modulated
signal are given by a(t) = +X∞ k=−∞ kAkk rect[0,T ](t − kT ) (196) ϕ(t) = +X∞ k=−∞ π 2 (1− sign (Ak)) rect[0,T ](t − kT ) (197)
Note the presence of the time windowing function
Power spectral density of the ASK-2
Hypothesis: both signals ±A have an equal probability!
I The mean µA of the random variable Ak is equal to 0.
I The variance is given by σA2 = E A2k = A2.
I p(t) = rect[0,T ](t). Thus, its Fourier transform is
P(f ) = e−j2πf T2 T sinc(fT ) (198)
Therefore, the power spectrum of the complex envelop is
γes(f ) = kP(f )kT 2 hσA2 +kµAk2P+m=∞−∞ T1δ f − mT i : γes(f ) = A 2T sinc2(fT ) (199)
and that of the ASK-2 modulated signal is
γs(f ) = A
2T sinc2[(f − f
c) T ] + sinc2[(f + fc) T ]
4 (200)
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ASK-2: bandwidth and spectral efficiency
The PSD of ASK-2 is γs(f ) = A 2T sinc2[(f − f c) T ] + sinc2[(f + fc) T ] 4 (201) At fixed bitrate Rb:
Technique bandwidth spectral efficiency η Baseband (NRZ) W = 0.6 Rb η = 0.6 RRb b ' 1.6
Constellation or state diagram: definition I
We have:
s(t) = sI(t) cos (2πfct)−sQ(t) sin (2πfct) (202) and, for the complex envelope:
es(t) = sI(t)+ jsQ(t) (203) The alternative is s(t) = as(t)cos (2πfct +φs(t)) (204) with as(t) = q s2 I (t) + sQ2(t) (205) and φs(t) = tan−1 sQ(t) sI(t) (206)
Definition (Constellation diagram)
The plot of es(t) in a diagram whose axis units are (cos (2πfct),
− sin (2πfct)) defines the constellation diagram.
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Constellation or state diagram: definition II
Constellation diagram of a ASK-2 or BPSK modulation
(A, 0) (−A, 0)
−p(t) sin (2πfct + ϕc)
p(t) cos (2πfct + ϕc)
The presence of p(t) reinforces the idea that there is only one symbol per time slot, but it is often dropped. Likewise, ϕc is often taken equal to 0.
Constellation diagram: purposes
(A, 0) (−A, 0)
−p(t) sin (2πfct + ϕc)
p(t) cos (2πfct + ϕc)
I Representation of the possible states in the complex plane.
I See how the state diagram is used (here, we immediately see that the sin() axis is not used).
I The distance between states is essential for finding the Pe.
Closer states mean less resistance to noise.
I See the paths from one state to another (trajectories).
Note that we move from one state to another state at the rhythm of the symbol rate (not the bit rate!)
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Constellation diagram of an On-Off Shift Keying (OOK)
(A, 0)
−p(t) sin (2πfct + ϕc)
p(t) cos (2πfct + ϕc) (0, 0)
Infrared signals are usually sent using On-Off Shift Keying (because it is hard to determine the phase of an infrared signal).
Constellation diagrams
OOK, ASK-2≡BPSK≡PSK-2, and QPSK.
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